I don't know weather it is possible for encoder to do such an analysis of a source audio, but it would be great it yes.

It's only a matter of finding the right formula/algorithm.

x264 video encoder has encoding mode called Constant Rate Factor. In this mode number (16, 17, etc) is used to define desired quality (lower - better quality and higher bitrate), and encoder does not care about bitrate, only about keeping rate factor constant. It is a question, why nobody has invented something similar for audio encoding (except lossyWAV, which needs too much bitrate for acceptable quality)?----------------Opus 1.1 Alpha has some bugs, which can be found using samples from thread High Frequency Listening Test Samples. For example, at 16-24 kbps Opus gives this:and for 32-40 kbps it gives this:For samples 1_12kHz, 1_20kHz, 2_8kHz, 2_12kHz and 2_20kHz Opus sounds wrongly even at 512 kbps.Full set of files is here (problematic sampes are marked with exclamation mark). Hope, developers will use this samples in their work.

x264 video encoder has encoding mode called Constant Rate Factor. In this mode number (16, 17, etc) is used to define desired quality (lower - better quality and higher bitrate), and encoder does not care about bitrate, only about keeping rate factor constant. It is a question, why nobody has invented something similar for audio encoding (except lossyWAV, which needs too much bitrate for acceptable quality)?

I think every encoder with real vbr (not abr) does that? Lame has V(0-9), QT AAC has --tvbr (0-127), Vorbis has -q((-2)-10). The bitrate may vary a lot with these settings between different songs/genres.

Having a file with two different quality levels is not what VBR is meant to do. I think to have that work you'd have to have some kind of filter or processing that attempted to classify the signal as audio or music and then adjusted the encoder's parameters from frame to frame. Its probably not too hard to do, but its also a very strange thing to want to implement so maybe no one has done so.

Right, I see now that DonP didn’t encode a single track with three parts, but three separate tracks. That’s not what softrunner was asking about, then. But DonP’s results and possible explanation still have a good degree of relevance.

My previous posts were written under the assumption that we were talking about large regions of broadly differing complexity/amplitude/whatever in the same file encoded at a single quality, such as speech and music. Barring some other aspect of speech that makes the encoder think it’s similarly complex to music, I would have expected a sizeable difference in bitrate between the two sections.

QUOTE (saratoga @ Feb 14 2013, 20:22)

VBR gives you constant quality.

Having a file with two different quality levels is not what VBR is meant to do. I think to have that work you'd have to have some kind of filter or processing that attempted to classify the signal as audio or music and then adjusted the encoder's parameters from frame to frame. Its probably not too hard to do, but its also a very strange thing to want to implement so maybe no one has done so.

As above, what I think softrunner was asking about, and certainly what I was talking about, was a single file with two different parts and the possibility for an encoder to provide significantly differing bitrates if the two parts differ in complexity. Again, perhaps my presumption that there should be a big difference was incorrect. I have no experience with the specific scenario and vanishingly small experience with encoded speech.